| .. |
|
adaptation
|
Apply resolution-bitrate limits collected from field trial (cl/294600) for AV1.
|
2023-03-16 19:04:32 +00:00 |
|
test
|
pc: Add asynchronous RtpSender::SetParameters() call
|
2022-11-15 15:31:40 +00:00 |
|
audio_receive_stream.cc
|
Rename AudioReceiveStream to AudioReceiveStreamInterface
|
2022-05-23 08:44:26 +00:00 |
|
audio_receive_stream.h
|
stats: use uint64_t for RTCSentRtpStreamStats.packetsSent
|
2023-03-16 06:46:19 +00:00 |
|
audio_send_stream.cc
|
Reland "Wire up non-sender RTT for audio, and implement related standardized stats."
|
2021-09-06 14:26:55 +00:00 |
|
audio_send_stream.h
|
pc: Add asynchronous RtpSender::SetParameters() call
|
2022-11-15 15:31:40 +00:00 |
|
audio_sender.h
|
Refactoring AudioSender api out of AudioSendStream for more abstraction to reuse AudioTransportImpl for voip api
|
2020-01-13 18:31:30 +00:00 |
|
audio_state.cc
|
Remove chromium clang style errors affecting sdk/android/media_jni
|
2018-04-09 13:55:49 +00:00 |
|
audio_state.h
|
Async audio processing API
|
2020-10-02 12:33:34 +00:00 |
|
bitrate_allocator.cc
|
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 1
|
2022-03-09 13:23:21 +00:00 |
|
bitrate_allocator.h
|
Use backticks not vertical bars to denote variables in comments for /call
|
2021-07-27 18:29:33 +00:00 |
|
bitrate_allocator_unittest.cc
|
Adopt absl::string_view in call/
|
2022-05-17 12:00:45 +00:00 |
|
bitrate_estimator_tests.cc
|
Remove rtp header extension from config of Call audio and video receivers
|
2023-01-31 11:58:43 +00:00 |
|
BUILD.gn
|
Ensure FakeNetwork propages arrival_time
|
2023-01-19 15:27:33 +00:00 |
|
call.cc
|
Remove rtp header extension from config of Call audio and video receivers
|
2023-01-31 11:58:43 +00:00 |
|
call.h
|
Delete Call dependency on ProcessThread as unused
|
2022-06-21 08:59:38 +00:00 |
|
call_config.cc
|
Add parameter to control the pacer's burst outside of field trials.
|
2022-11-15 08:46:30 +00:00 |
|
call_config.h
|
Add parameter to control the pacer's burst outside of field trials.
|
2022-11-15 08:46:30 +00:00 |
|
call_factory.cc
|
Remove CoDel from webrtc::SimulatedNetwork.
|
2022-09-08 06:51:05 +00:00 |
|
call_factory.h
|
Delete Call dependency on ProcessThread as unused
|
2022-06-21 08:59:38 +00:00 |
|
call_perf_tests.cc
|
Ensure CallTest derived tests per default set min/max audio bitrate.
|
2023-01-26 17:36:01 +00:00 |
|
call_unittest.cc
|
Delete Call dependency on ProcessThread as unused
|
2022-06-21 08:59:38 +00:00 |
|
degraded_call.cc
|
Reland "Delete PacketReceiver::DeliverPacket from all implementations"
|
2023-01-25 18:18:29 +00:00 |
|
degraded_call.h
|
Reland "Delete PacketReceiver::DeliverPacket from all implementations"
|
2023-01-25 18:18:29 +00:00 |
|
DEPS
|
SimulcastEncoderAdapter: Use FramerateController instead of FramerateControllerDeprecated.
|
2021-08-30 10:20:55 +00:00 |
|
fake_network_pipe.cc
|
Reland "Delete PacketReceiver::DeliverPacket from all implementations"
|
2023-01-25 18:18:29 +00:00 |
|
fake_network_pipe.h
|
Reland "Delete PacketReceiver::DeliverPacket from all implementations"
|
2023-01-25 18:18:29 +00:00 |
|
fake_network_pipe_unittest.cc
|
Reland "Delete PacketReceiver::DeliverPacket from all implementations"
|
2023-01-25 18:18:29 +00:00 |
|
flexfec_receive_stream.cc
|
[Cleanup] Add missing #include. Remove useless ones.
|
2018-10-23 11:32:56 +00:00 |
|
flexfec_receive_stream.h
|
Add SetPayloadType to FlexfecReceiveStream.
|
2022-07-28 21:24:50 +00:00 |
|
flexfec_receive_stream_impl.cc
|
Remove rtp header extension from config of Call audio and video receivers
|
2023-01-31 11:58:43 +00:00 |
|
flexfec_receive_stream_impl.h
|
Remove rtp header extension from config of Call audio and video receivers
|
2023-01-31 11:58:43 +00:00 |
|
flexfec_receive_stream_unittest.cc
|
Remove rtp header extension from config of Call audio and video receivers
|
2023-01-31 11:58:43 +00:00 |
|
OWNERS
|
Update OWNERS for call/
|
2022-06-03 12:01:46 +00:00 |
|
packet_receiver.h
|
Reland "Delete PacketReceiver::DeliverPacket from all implementations"
|
2023-01-25 18:18:29 +00:00 |
|
rampup_tests.cc
|
Stop overriding extensions in rampup tests
|
2023-01-25 13:18:49 +00:00 |
|
rampup_tests.h
|
Stop overriding extensions in rampup tests
|
2023-01-25 13:18:49 +00:00 |
|
receive_stream.h
|
Remove rtp header extension from config of Call audio and video receivers
|
2023-01-31 11:58:43 +00:00 |
|
receive_time_calculator.cc
|
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf
|
2022-03-29 10:14:00 +00:00 |
|
receive_time_calculator.h
|
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf
|
2022-03-29 10:14:00 +00:00 |
|
receive_time_calculator_unittest.cc
|
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 2
|
2022-03-09 22:17:52 +00:00 |
|
rtp_bitrate_configurator.cc
|
Allow setting a bandwidth cap for relayed connections.
|
2020-03-26 20:41:46 +00:00 |
|
rtp_bitrate_configurator.h
|
Remove RTC_DISALLOW_COPY_AND_ASSIGN more.
|
2022-01-20 11:00:18 +00:00 |
|
rtp_bitrate_configurator_unittest.cc
|
Revert "In RtpBitrateConfigurator ignore new parameters when set to default values."
|
2020-01-10 16:39:51 +00:00 |
|
rtp_config.cc
|
Prepare to rename RTC_NOTREACHED to RTC_DCHECK_NOTREACHED
|
2021-11-15 21:44:59 +00:00 |
|
rtp_config.h
|
Update old TODO comments
|
2022-07-05 09:59:33 +00:00 |
|
rtp_demuxer.cc
|
Adopt absl::string_view in call/
|
2022-05-17 12:00:45 +00:00 |
|
rtp_demuxer.h
|
Adopt absl::string_view in call/
|
2022-05-17 12:00:45 +00:00 |
|
rtp_demuxer_unittest.cc
|
Adopt absl::string_view in call/
|
2022-05-17 12:00:45 +00:00 |
|
rtp_packet_sink_interface.h
|
|
|
|
rtp_payload_params.cc
|
Reland "Make SimulcastIndex() and SpatialIndex() distinct (remove fallback)."
|
2023-02-21 18:30:35 +00:00 |
|
rtp_payload_params.h
|
For VP9 assume max number of spatial layers to simulate generic descriptor
|
2022-06-08 11:36:54 +00:00 |
|
rtp_payload_params_unittest.cc
|
Introduce EncodedImage.SimulcastIndex().
|
2023-02-15 15:02:57 +00:00 |
|
rtp_stream_receiver_controller.cc
|
Change RecoveredPacket::OnRecoveredPacket to produce webrtc::RtpPacketReceived
|
2022-12-22 14:04:21 +00:00 |
|
rtp_stream_receiver_controller.h
|
Change RecoveredPacket::OnRecoveredPacket to produce webrtc::RtpPacketReceived
|
2022-12-22 14:04:21 +00:00 |
|
rtp_stream_receiver_controller_interface.h
|
Demote RtpStreamReceiverController AddSink/RemoveSink to private
|
2022-07-06 09:31:54 +00:00 |
|
rtp_transport_config.h
|
Add parameter to control the pacer's burst outside of field trials.
|
2022-11-15 08:46:30 +00:00 |
|
rtp_transport_controller_send.cc
|
Stop Posting tasks when we don't need to.
|
2023-03-06 15:13:39 +00:00 |
|
rtp_transport_controller_send.h
|
Stop Posting tasks when we don't need to.
|
2023-03-06 15:13:39 +00:00 |
|
rtp_transport_controller_send_factory.h
|
Refactor some config plumbing in call/.
|
2022-11-16 09:18:40 +00:00 |
|
rtp_transport_controller_send_factory_interface.h
|
Delete Call dependency on ProcessThread as unused
|
2022-06-21 08:59:38 +00:00 |
|
rtp_transport_controller_send_interface.h
|
Replace TaskQueue with MaybeWorkerThread in RtpTransportControllerInterface
|
2022-10-10 11:56:52 +00:00 |
|
rtp_video_sender.cc
|
Add a field trial string to make enable_retransmit_all_layers configurable.
|
2023-03-03 15:20:41 +00:00 |
|
rtp_video_sender.h
|
Reland "Remame VideoSendStream::UpdateActiveSimulcastLayers to StartPerRtpStream"
|
2022-12-02 12:03:25 +00:00 |
|
rtp_video_sender_interface.h
|
Reland "Remame VideoSendStream::UpdateActiveSimulcastLayers to StartPerRtpStream"
|
2022-12-02 12:03:25 +00:00 |
|
rtp_video_sender_unittest.cc
|
Introduce EncodedImage.SimulcastIndex().
|
2023-02-15 15:02:57 +00:00 |
|
rtx_receive_stream.cc
|
Updated associated payload types without recreating receive streams.
|
2022-08-16 13:38:24 +00:00 |
|
rtx_receive_stream.h
|
Updated associated payload types without recreating receive streams.
|
2022-08-16 13:38:24 +00:00 |
|
rtx_receive_stream_unittest.cc
|
Store RtpPacketReceived::arrival_time as Timestamp.
|
2021-05-05 16:22:33 +00:00 |
|
simulated_network.cc
|
Reland "Add documentation, tests and simplify webrtc::SimulatedNetwork."
|
2022-11-06 13:14:26 +00:00 |
|
simulated_network.h
|
Reland "Add documentation, tests and simplify webrtc::SimulatedNetwork."
|
2022-11-06 13:14:26 +00:00 |
|
simulated_network_unittest.cc
|
Reland "Add documentation, tests and simplify webrtc::SimulatedNetwork."
|
2022-11-06 13:14:26 +00:00 |
|
simulated_packet_receiver.h
|
Calculate next process time in simulated network.
|
2019-02-08 19:33:17 +00:00 |
|
syncable.cc
|
|
|
|
syncable.h
|
Rename AudioReceiveStream to AudioReceiveStreamInterface
|
2022-05-23 08:44:26 +00:00 |
|
version.cc
|
Update WebRTC code version (2023-03-29T04:08:24).
|
2023-03-29 06:05:37 +00:00 |
|
version.h
|
Add WebRTC code freshness version string.
|
2020-12-14 16:22:35 +00:00 |
|
video_receive_stream.cc
|
Remove rtp header extension from config of Call audio and video receivers
|
2023-01-31 11:58:43 +00:00 |
|
video_receive_stream.h
|
Allow RTX ssrc to be updated on receive streams
|
2023-02-01 12:54:46 +00:00 |
|
video_send_stream.cc
|
Change the type of RTCVideoSourceStats.framesPerSecond
|
2021-11-16 11:21:41 +00:00 |
|
video_send_stream.h
|
Add scalability mode to RTCOutboundRtpStreamStats stats
|
2022-12-08 11:46:06 +00:00 |
ae049f1924